U.S. patent application number 10/159600 was filed with the patent office on 2003-12-04 for frequency compander for a telephone line.
Invention is credited to Baxter, Kevin Cotton, Fisher, Ken Scott, Holmes, Fred H..
Application Number | 20030225583 10/159600 |
Document ID | / |
Family ID | 29582959 |
Filed Date | 2003-12-04 |
United States Patent
Application |
20030225583 |
Kind Code |
A1 |
Fisher, Ken Scott ; et
al. |
December 4, 2003 |
Frequency compander for a telephone line
Abstract
A frequency compander for improving the frequency response of a
telephone line when used for remote broadcasting. The inventive
device comprises an encoder for compressing the frequency spectrum
of an audio signal and a decoder for expanding the signal back to
its original spectrum. Preferably the encoder comprises: an
anti-aliasing filter; an A/D converter for digitizing incoming
audio; a DSP for compressing the audio; and a D/A converter for
outputting compressed audio to the phone line. The decoder
comprises: an anti-aliasing filter; an A/D converter for digitizing
the incoming compressed signal; a DSP for restoring the original
audio; and a D/A converter for outputting program audio. In a
preferred embodiment, encoding and decoding are performed in the
frequency domain. In another preferred embodiment, encoding and
decoding are performed in the time domain using trigonometric
transformations.
Inventors: |
Fisher, Ken Scott; (Los
Angeles, CA) ; Baxter, Kevin Cotton; (Santa Clara,
CA) ; Holmes, Fred H.; (Cleveland, OK) |
Correspondence
Address: |
KEN FISHER
5521 CLEON AVE.
NORTH HOLLYWOOD
CA
91601
US
|
Family ID: |
29582959 |
Appl. No.: |
10/159600 |
Filed: |
May 31, 2002 |
Current U.S.
Class: |
704/500 ;
704/E19.01 |
Current CPC
Class: |
G10L 19/02 20130101;
G10L 19/093 20130101 |
Class at
Publication: |
704/500 |
International
Class: |
G10L 021/00 |
Claims
What is claimed is:
1. A frequency compander for improving the bandwidth of audio sent
via a public network comprising: input means for receiving an audio
signal; encoding means for compressing the frequency spectrum of
said audio signal, said encoding means having an output means for
outputting a compressed audio signal; and network interface means
for connection to a public network, wherein said compressed audio
is transmitted to said public network through said network
interface means.
2. The frequency compander of claim 1 wherein said encoding means
comprises a digital signal processor, said input means comprises an
analog to digital converter, and said output means comprises a
digital to analog converter.
3. The frequency compander of claim 2 wherein said encoding means
further comprises a software program for performing an FFT and an
inverse FFT.
4. The frequency compander of claim 1 wherein said input means is a
first input means and said output means is a first output means,
further comprising: a second input means for inputting compressed
audio received from said network interface; a decoding means in
communication with said second input means for expanding said
compressed audio; and a second output means for delivering program
audio, wherein, said program audio is expanded from said compressed
audio.
5. A frequency compander for improving the frequency response of an
audio transmission channel comprising: an anti-aliasing filter
having an input for receiving an audio signal; an analog to digital
converter in communication with said anti-aliasing filter to
digitize said audio signal; a digital signal processor in
communication with said analog to digital converter, said digital
signal processor executing a computer program which includes steps
to compress the frequency spectrum of said audio signal; a digital
to analog converter for outputting compressed audio from said
digital signal processor.
6. The frequency compander of claim 5 further wherein said analog
to digital converter is a first analog to digital converter, said
input is a first input, and said digital to analog converter is a
first digital to analog converter, further comprising: a second
analog to digital converter having a second input for inputting a
compressed audio signal; a second digital to analog converter for
outputting an expanded audio, wherein said computer program further
includes steps to expand said compressed audio received at said
second analog to digital and output expanded audio at said second
digital to analog converter.
7. A method for compressing audio information including the steps
of: (a) inputting an audio signal; (b) digitizing said audio
signal; (c) compressing the frequency spectrum from the digitized
audio signal of step (b) into compressed data; (d) converting said
compressed data to an analog form; (e) repeating steps (b)-(d) on a
periodic basis.
8. The method for compressing audio information of claim 7 wherein
step (c) includes the steps of: (c)(i) performing a fast Fourier
transform on the digitized audio signal of step (b) to form a
frequency domain table; (c)(ii) increasing the size of said
frequency domain table, in proportion to the degree of frequency
compression to be performed, the new table locations being disposed
above the existing data in said frequency domain table, relative to
the spectral content of said existing data, said new locations
being cleared; and (c)(iii) performing an inverse fast Fourier
transform on said frequency domain table of increased size of step
(c)(ii);
9. The method for compressing audio information of step 7 wherein
the compressing of step (c) comprises a trigonometric
transformation.
10. A method for expanding the frequency spectrum of a compressed
audio signal including the steps of: (a) inputting a compressed
audio signal; (b) digitizing said compressed audio signal; (c)
expanding the frequency spectrum from the digitized compressed
audio signal of step (b) into program audio data; (d) converting
said program audio data to an analog form; (e) repeating steps
(b)-(d) on a periodic basis.
11. The method for expanding the frequency spectrum of a compressed
audio signal of claim 10 wherein step (c) includes the substeps of:
(c)(i) performing a fast Fourier transform on the digitized
compressed audio signal of step (b) to form a frequency domain
table, said frequency domain of a size to include spectral
information of said compressed audio signal at least to the highest
frequency to be recovered; (c)(ii) decreasing the size of the table
to contain only spectral information from 0 Hz a first frequency,
said first frequency being the highest frequency programmed in said
compressed audio data, discarding the information stored in said
table for frequencies above said first frequency; and (c)(iii)
performing an inverse fast Fourier transform on said frequency
domain table of decreased size of step (c)(ii);
12. The method for expanding the frequency spectrum of a compressed
audio signal of claim 10 wherein the expanding of step (c)
comprises a trigonometric transformation.
13. A method for transmitting audio information between a first
point and a second point over a public network connection such that
the transmitted audio will be received with a spectral content
greater than the frequency response of the public network
connection, including the steps of: (a) connecting a first
frequency compander to a public network at the first point; (b)
connecting a second frequency compander to said public network at
the second point; (c) making a connection between said first
frequency compander and said second frequency compander on said
public network; (d) providing an audio program to said first
frequency compander for compressed transmission on said public
network; (e) compressing said audio program in said first frequency
compander into a compressed audio program; (f) transmitting said
compressed audio program on said public network; (g) receiving said
compressed audio program at said second frequency compander; (h)
expanding said compressed audio program in said second frequency
compander into a restored audio program; (i) outputting said
restored audio program from said second frequency compander.
14. A method for selecting a decoding scheme in a frequency
compander including the steps of: (a) connecting a frequency
compander to a telephone line at a first location; (b) connecting a
remote broadcast device to a telephone line at a second location;
(c) establishing a connection between said remote broadcast device
and said frequency compander over the telephone network; (d)
transmitting a test tone of a predetermined frequency from said
remote broadcast device to said frequency compander; (e)
determining the frequency of the tone received at said frequency
compander; and (f) selecting a mode of operation based on the
frequency determined in step (e) from the group consisting of:
(f)(i) frequency extender mode; (f)(ii) frequency companding with
shifting mode; (f)(iii) frequency companding without shifting
mode.
15. The method for selecting a decoding scheme in a frequency
compander of claim 14 including the addition steps of: (g) upon
selecting the operating mode of (f)(ii), subtracting said
predetermined frequency from said the frequency of said tone
received; and (h) adjusting the shift frequency to the difference
determined in step (g).
16. A precision frequency extender for extending the lower
frequency range by shifting the frequency of an audio program
comprising: an A/D converter for digitizing incoming audio; a
digital signal processor, said digital signal processor receiving
digitized audio from said A/D converter; a D/A converter in
communication with said digital signal processor for outputting
frequency shifted audio, wherein said digital signal processor
performs a series of programming steps to shift the frequency
spectrum of said incoming audio according to a trigonometric
transformation to create said frequency shifted audio and outputs
said frequency shifted audio via said D/A converter.
17. The precision frequency extender of claim 16 wherein the
frequency extender is an encoder and wherein said digital signal
processor shifts the frequency spectrum of said incoming audio up
250 Hz.
18. The precision frequency extender of claim 16 wherein the
frequency extender is a decoder and wherein said digital signal
processor shifts the frequency spectrum of said incoming audio down
by 250 Hz.
Description
BACKGROUND OF THE INVENTION
[0001] 1. Field of the Invention
[0002] The present invention relates to frequency extenders for a
telephone line. More particularly, but not by way of limitation,
the present invention relates to a frequency extender to expand the
bandwidth of a dialup telephone line used to carry remote audio
programming.
[0003] 2. Background of the Invention
[0004] Virtually every broadcaster, whether radio or television,
has at some point in time, felt the need to carry programming
originating from a remote location. In response to this need, a
number of solutions have been developed. Unfortunately, every
method presently used for remote broadcasting suffers from its own
set of disadvantages.
[0005] Presently radio frequency devices are the favored method for
sending programming from a remote location to a studio or
transmitter for broadcast. Devices offered for this purpose are
often referred to as a "remote pickup unit" or "RPU."
[0006] Perhaps the favored RPU is a microwave link. Such systems
have excellent bandwidth, good signal to noise performance, and
usually include bi-directional operation. In most cases the
microwave RPU is built into a van, SUV, truck, or the like. Since
microwave signals are basically line-of-sight in nature, there is
normally an extendible mast on the vehicle to raise the antenna
high enough to clear obstacles and increase the range. Even so,
microwave links have a limited range. In addition to line of sight
operation, microwave systems suffer from a number of other
limitations which include: the equipment is expensive, so
expensive, in fact, that most small market radio stations would be
hard pressed to purchase even a single system; there is setup time
in extending the mast and aiming the remote antenna towards the
receiving antenna; microwave systems require a dedicated vehicle;
overhead power lines can pose a significant risk to the operator
while extending the mast; and, like all RF devices, there is a
potential for interference and fade.
[0007] Perhaps the most pervasive RPU is the UHF or VHF two-way
radio. While two way radios are available for a number of bands, by
far UHF radios are the most popular, typically operating in the
vicinity of 450 MHz. These radios offer moderate bandwidth and cost
a mere fraction of the cost of microwave systems. Unfortunately,
two-ways are particularly subject to interference, especially in
large metropolitan areas where the frequency selected by a radio
station for its two-way equipment is likely shared with other
businesses. As a result, a remote broadcast may be interrupted by
other radio operators. Even if a broadcaster's two-way radio
frequency is exclusive, use of such radios has become so pervasive
that interference from equipment operating on adjacent channels is
common place. Furthermore, while two-way radio transmissions are
not limited to line of sight like their microwave counterparts,
such radios still suffer from limited range and require a
significant investment by a broadcaster.
[0008] Remote programming may also be sent to a radio station over
the public telephone network. A telephone link has virtually
unlimited range, is rarely affected by outside noise sources, and
requires only a minimal investment. Unfortunately, if a switched
line is used, the bandwidth provided by a telephone connection is
marginal at best. The frequency response of a telephone line is
generally 300 Hz to 3100 Hz. In comparison, the frequency response
of an FM radio broadcast is generally 30 Hz to 15 KHz. Audio sent
through a phone line is degraded to the point where even the most
untrained ear can distinguish it from other programming. In fact,
in competitive radio markets some broadcasters refuse to use dialup
phone lines to carry any programming, even for live remotes.
[0009] Since bandwidth is the principal disadvantage to using the
switched telephone network, a number of techniques are used by
radio stations to reduce the problem of limited bandwidth. One
solution is to employ a dedicated leased telephone line. Leased
lines are directly connected between the source and destination
locations. While 10 KHz bandwidth may be available with such lines,
the costs are substantially higher than with a conventional phone
line, the phone company requires some lead time to install and
connect the line, and there is usually a minimum period over which
the line must be leased. As a result, a leased line is not
practical for most remote broadcasting events.
[0010] Another solution to the bandwidth problem is the frequency
extender. In its simplest form, a frequency extender shifts the
source audio up 250 Hz prior to its transmission over the phone
lines. At the receiving end, the frequency of the program audio is
shifted back down 250 Hz to its original frequency. The magic of a
frequency extender lies in the nature of the frequency range
provided by the telephone company on a phone line. As previously
mentioned, the typical bandwidth of a phone line is 300 Hz to 3100
Hz, a range of just over three octaves. The frequency shifting
technique used by a frequency extender shifts the frequency range
to roughly 50 Hz to 2850 Hz, or over five and one-half octaves. At
the upper end, where frequency range is sacrificed, 250 Hz is a
mere fraction of an octave. At the lower end, the added range from
50 Hz to 300 Hz is well over two octaves. As those familiar with
such devices will readily appreciate, as a result of frequency
extension, the audio exhibits a fuller, richer sound than audio
transmitted without the benefit of such extension. Of course, even
with the improved sound, the high end of the audio spectrum is
still absent from the program.
[0011] To improve high-end performance, multi-line extenders are
available. These devices use this same frequency-shifting technique
to recover higher portions of the audio spectrum, 2800 Hz at a
time. Beyond the obvious problems of requiring the simultaneous use
of multiple telephone lines, these devices traditionally have
required some setup to compensate for variances in the
characteristics of each of the phone lines.
[0012] More recently, the broadcast industry has turned to digital
codecs. Codecs are available for conventional phone lines, ISDN
lines, and even for use over the Internet. In a digital codec,
program audio is first digitized, then radically compressed,
transmitted in digital form by a modem across the telephone
network, received by a modem at the receiving end, decompressed,
and finally, converted back to analog form. Such devices can yield
amazing improvements in the apparent bandwidth. Unfortunately, they
also have a number of limitations, including: 1) digital codecs are
presently very expensive, at least compared to their
frequency-shifting counterparts; 2) the actual digital throughput
of a particular connection is unpredictable and can vary widely,
not only from connection-to-connection between the same two
locations, but even during a single session; 3) the reproduced
audio is typically reconstructed through a "model" and is not the
actual audio produced so that the result may include spurious
sounds not in the original audio, sounds may be lost in the
conversion process, and downstream processing of the audio can
yield unpredictable and unwanted results; 4) the quality of the
audio is dependent on the digital throughput; and 5) long gaps in
the program audio can occur if the modems lose synchronization and
must re-handshake. Despite the popularity of codecs, the state of
the art of digital transmission over the switched telephone network
is just not quite ready for audio broadcast purposes.
[0013] Yet another method for handling a remote broadcast is via a
cellular telephone connection. While a cellular-to-cellular
connection is possible, normally a cellular telephone is used to
call a conventional dialup line at the radio station. Analog cell
phones are rapidly becoming a relic. However, at least as long as
signal strength is adequate, the problems encountered with a
cellular connection are basically the same as those encountered
with a conventional telephone line, specifically bandwidth. Like a
conventional connection, this problem may be somewhat relieved
through the use of frequency extenders. An additional annoyance
with analog cell phones is the occasional switching between cell
sites which causes a momentary "hole" in the audio signal.
[0014] Presently, the cellular network is transitioning to all
digital. Like the digital frequency extender mentioned above,
digital cell phones rely heavily on compression techniques to
maximize the amount of audio information which can be transmitted
at a relatively low bit rate. Unfortunately, these compression
techniques produce a received signal which is essentially a
synthesis of the original signal. As is well known in the art, as
the system becomes congested or as signal strength degrades, the
recovered audio often becomes unintelligible. Furthermore,
downstream processing of audio transmitted over a digital cellular
connection may produce unpredictable results. Present frequency
compression technique are generally not well suited for use with
digital cellular phones.
[0015] It should be noted that many digital cell phones provide a
data connection and there are devices which make use of such a
connection to transmit compressed and digitized audio via the
digital port on the cell phone. Presently the data rates provided
through such phones is too low for the transmission of audio
information, even when heavily processed, especially in light of
the fact that with many phones, the digital connection may be
shared among several users, i.e. with a CDPD connection.
[0016] Finally, it is a common practice in the field to direct
talent over a separate communication channel typically know as an
"interruptible feedback" line or "IFB." Particularly in the
television industry, a phone connection, or cell phone, is often
used for an IFB even when programming is sent via an RF link. Since
the talent receives cues over the IFB, it is important that such
cues be readily intelligible. Thus there is a need for systems
which will improve the quality of off-line audio used for remote
cuing.
[0017] Thus it is an object of the present invention to provide a
system and method for frequency extension which provides suitable
bandwidth over a conventional switched telephone connection.
[0018] It is a further object of the present invention to transmit
the information in an audio form such that consistent results are
provided from one connection to the next.
[0019] It is still a further object of the present invention to
provide a lowcost frequency extender which substantially doubles
the bandwidth of a telephone connection.
SUMMARY OF THE INVENTION
[0020] The present invention provides a frequency compander for
connection to a telephone line, or a cellular telephone network,
which will provide a substantial improvement in bandwidth of the
telephone line. Unlike prior art extenders which merely shift the
frequency to make better use of the available bandwidth, the
present invention sacrifices signal-to-noise performance of the
connection in exchange for increased bandwidth.
[0021] In a preferred embodiment, an encoder processes program
audio by filtering the signal, converting the audio to a digital
form, and compressing the audio into a narrower spectrum through a
process described herein as "frequency companding". In general, the
term "companding" is used to describe a combined process of
COMPressing and exPANDing (emphasized with capital letter to
improve clarity). In one preferred embodiment, the signal is
transformed into the frequency domain through a continuous Fourier
Transform. The transformed data is manipulated to maintain the
resolution of the transformed data but to compress the information
into one-half, or less, of the spectrum. A continuous inverse
transform is then performed and the signal is converted back to
analog to for transmission over the public network. At the
receiving end, the process is reversed in a decoder to expand the
signal, in the frequency domain, back to the original program.
[0022] The companding process is not without its costs, the
signal-to-noise ratio of the original signal suffers degradation
due to phase noise arising in the companding process and through
lost resolution in the noise floor of the signal. In return,
however, the decoded signal is produced with roughly twice the
bandwidth, or more, of the public network channel used. It is
generally reasonable to expect -45 dB, or better, signal to noise
ratio on a dialup line. With frequency doubling, the signal will
still have about -40 dB signal to noise ratio.
[0023] In a second preferred embodiment, the frequency is
compressed into at least half the spectrum, in a point-by-point
process using a well-known trigonometric transformation. At the
decoder, the signal is expanded using an inverse trigonometric
transformation.
[0024] In another preferred embodiment the inventive frequency
compander includes a microphone input, a headphone output, and a
keypad for management of the public network connection such that
the device is a stand alone system for performing a remote
broadcast.
[0025] The present invention is distinguishable from prior art
systems in that: 1) analog frequency extenders only shift the
frequency of the program audio, as opposed to compressing, to
restore the missing lower frequencies; and 2) present digital
frequency extenders compress the audio and attempt transmission in
a digital form, as opposed to sending an analog audio signal
shifted down one or more octaves, which relies on modeling of the
human hearing or vocal tract to decompress. The advantage of the
present invention over analog frequency extenders is a vast
improvement in bandwidth. Advantages of the present invention over
prior art digital extenders include: dramatically lower cost; more
consistent operation, e.g., less dependency on the quality of the
phone line for the quality of the received audio; and an analog
output which is suitable for downstream processing.
[0026] Further objects, features, and advantages of the present
invention will be apparent to those skilled in the art upon
examining the accompanying drawings and upon reading the following
description of the preferred embodiments.
BRIEF DESCRIPTION OF THE DRAWINGS
[0027] FIG. 1 provides a flow diagram for a process for encoding
frequency extended audio through an FFT.
[0028] FIG. 2 provides a flow diagram for a process for decoding
frequency extended audio through an FFT.
[0029] FIG. 3 provides a flow diagram for a process for encoding
frequency extended audio through a trigonometric transform.
[0030] FIG. 4 provides a flow diagram for a process for decoding
frequency extended audio through a trigonometric transform.
[0031] FIG. 5 provides a perspective view of the inventive
frequency compander.
[0032] FIG. 6 provides a diagram of a system for remote broadcast
incorporating the inventive frequency compander.
[0033] FIG. 7 provides a block diagram of the circuitry of a
preferred frequency compander.
DESCRIPTION OF THE PREFERRED EMBODIMENTS
[0034] Before explaining the present invention in detail, it is
important to understand that the invention is not limited in its
application to the details of the construction illustrated and the
steps described herein. The invention is capable of other
embodiments and of being practiced or carried out in a variety of
ways. It is to be understood that the phraseology and terminology
employed herein is for the purpose of description and not of
limitation.
[0035] Referring now to the drawings, wherein like reference
numerals indicate the same parts throughout the several views, a
typical frequency compander 500 is shown in FIG. 5. Preferably,
compander 500 comprises: enclosure 502; microphone jack 504,
typically an industry standard 3-pin XLR connector for the
connection of a microphone 602 (FIG. 6), or other audio source; a
headphone jack 506, typically a 1/4 inch phone jack for the
connection of a pair of headphones 604 (FIG. 6); a knob 508 for
adjusting the volume of the audio sent to headphones 604; and a
keypad 510 for controlling the operation of extender 500,
particularly with respect to its connection with a telephone
network.
[0036] In addition, compander 500 includes a modular phone jack
(not shown) for connection to a telephone network and a power
connector 704 for receiving electrical power on its rear panel (not
shown).
[0037] As discussed above purpose of frequency compander 500 is to
improve the fidelity of audio transmitted over a public network.
For purposes of this invention, a "public network" is a system for
point-to-point audio communication, such as, by way of example and
not limitation, the telephone network, a cellular phone/pcs
network, a two-way radio network, or the like. As also discussed
above, as used herein, the term "compander", or "companding," refer
to a device for, or the process of, frequency compressing and
frequency expanding.
[0038] A frequency compander is particularly useful for performing
a remote broadcast for a radio station, television station, etc.,
where because of the bandwidth normally broadcast by the station,
the listener has come to expect a level of sound quality better
than that normally available over the public networks. Frequency
companding is performed by encoding the audio signal at the remote
site by shifting the frequency of the signal, compressing the
spectrum occupied by the signal, or a combination of both,
transmitting the encoded signal over the network, and decoding
and/or shifting the compressed signal at the receiving end to
restore the original audio program.
[0039] Referring next to FIG. 7, circuitry for encoding and
decoding the audio signal 700 comprises: a digital signal processor
("DSP") 706; a microphone jack 506 for receiving an audio program;
an anti-aliasing filter 704 to low pass filter the audio at, or
below, one-half the sampling frequency to prevent quantitization
noise; a phone line interface 710 which provides phone line
functions such as, proper audio coupling to the phone line, 2
wire-to-4 wire conversion, ring detection hook management, etc.;
keypad 712 which allows the user to go off-hook, or on-hook, to
dial a phone number, or select operating modes of the extender;
potentiometer 714 for adjusting the volume of the audio delivered
to headphone connector 506.
[0040] With further reference to FIG. 1, wherein a flow diagram is
shown for the encoding process 100, audio is first brought to
compander 500 through connector 504 at step 102. As mentioned
above, the audio is directed through an anti-aliasing filter 704 at
step 104 to remove high frequency content above the maximum
frequency to be transmitted. Next at step 106, to encode the audio
program, DSP 706 performs a series of program steps which first
sample the incoming audio and convert the signal to digital form on
a periodic basis. At step 108, the incoming signal is transformed
from the time domain to the frequency domain on a sample-by-sample
basis through a conventional fast Fourier transform. Fourier
transforms are well known in the art and the programming of a DSP
to perform such a transform is well within the skill level of one
of ordinary skill in the art. To perform a continuous FFT on the
incoming data, a running buffer of the last sixteen samples are
used for each transformation. As each new sample is read, it is
placed at the beginning of the buffer while the oldest sample falls
off the opposite end of the buffer. As will be apparent to those
skilled in the art, the FFT produces a frequency domain table
wherein phase and amplitude information is stored relative to
frequency. Data stored in this table is indicative of
characteristics of the incoming signal relative to the spectral
content of the audio program. At step 110, the data is next copied
into the lower half of a table of twice the size of the original
table. Each location of the top half of both the larger table is
set to zero. Next, an inverse fast Fourier is performed on the
larger table on a sample-by-sample basis at step 112 to produce an
output buffer in the time domain wherein the spectral information
of the original signal is compressed by factor of two from the
original signal. Finally, the top value of the large table is
converted from digital to analog at step 114 to produce the audio
signal sent to the public network at 116.
[0041] Referring next to FIGS. 2 and 7, the process of decoding 200
is very similar in nature to the process of encoding 100 (FIG. 1).
First, at step 202, audio is received from the public network
interface 708. The audio is conditioned at step 204 by
anti-aliasing filter 704 to remove out-of-band noise received on
the phone line. The output of filter 704 is sampled, converted to
digital form, and placed in a 32-byte buffer in a first in first
out fashion at step 206. Next, at step 208, the buffer is
transformed to the frequency domain through a fast Fourier
transform. The lower half of the frequency domain table is then
copied into a table of one-half the size at step 210 before being
subjected to an inverse transform at step 212. The output buffer of
the transform of step 212 is 16-bytes in length and of the same
spectral content as the original signal at step 106 of the encoder
(FIG. 1), preferably on the order of twice that of the public
network. The top value of the buffer is then processed through a
digital to analog converter at step 214 to produce program audio at
step 216.
[0042] As will be apparent to those skilled in the art, if each
unit contains both encoding software and decoding software, then
high fidelity audio may be sent both from the remote location to
the studio and from the studio back to the remote location. This is
particularly helpful when a director at the studio wishes to cue
the talent at the remote location or where the program is sent back
to the remote location so that the talent may be cued
over-the-air.
[0043] Turning next to FIG. 6, a system for remote broadcasting 600
preferably comprises: a remote frequency compander 606 having an
audio source such as microphone 602 and a audio monitoring device
such as headphones 604; and a local frequency compander 612 located
at a studio or transmitter and connected to a public network,
typically a conventional dialup phone line 624. The audio output
620 of local compander 612 is preferably connected to an input of
mixer 618 so that incoming remote audio is under the control of
local personnel. Similarly, audio input 622 of local compander 612
is preferably connected to a monitor output of mixer 618 so that
audio returned to the remote location, i.e. audible directions or
actual on-the-air programming, is also under local control.
[0044] To initiate a remote broadcast, the operator connects remote
compander 606 to the phone network 624 and, using keypad 610, dials
the phone number of local compander 612. Upon detecting the ringing
signal, local compander 612 answers the call and a bi-directional
audio link is established. It should be noted that audio traveling
in both directions is compressed. Accordingly any reflections, or
echoes, caused by the phone network 624 will be properly
decompressed and thus sound normal either at headphones 604 or at
mixer 618. As will be appreciated by those who have attempted
uncompressed talk-back with analog extenders, both encoding and
decoding must be performed at both ends of the connection if
bi-directional communications are to be used.
[0045] Frequency companding can be accomplished in a number of
different ways. By way of example and not limitation, another
preferred method for frequency companding is shown in FIGS. 3 and
4, wherein well-known trigonometric transformations are used in
lieu of the FFT and inverse FFT steps 108-112 and 208-212 of FIGS.
1 and 2, respectively. In encoder 300, the audio information is
inputted at step 302, filtered at step 304, and converted to a
digital representation at periodic intervals at step 306, just as
in encoder 100 (FIG. 1). At step 308 frequency compression is then
performed on the sampled data on a sample-by-sample basis according
to the following equation:
cos(X/2)=sqrt(1/2+cos(X)/2)
[0046] where:
[0047] cos(X) is the audio input; and
[0048] cos(X/2) is the audio output.
[0049] It should be noted that the square root of the above
equation results in full-wave rectification of the output signal.
Accordingly, upon the detection of a local minimum value of the
input, a sign reversal of the output must be made. After this
adjustment, the result of this transformation is: frequency
shifting down one octave.
[0050] Following the transformation, the sample is converted back
to an analog signal at step 310 before being output to the public
network as compressed audio at step 312.
[0051] Like FFT decoder 200, trigonometric decoder 400 inputs
compressed audio from the public network at step 402, filters the
signal at step 404, and digitizes the signal at step 406.
Decompression is performed at step 408 using the inverse of the
transform of step 308 given by:
sin(2X)=2*sin(X)*cos(X)
[0052] where:
[0053] sin(2X) is the output of the decoder; and
[0054] sin(X) is the input to the decoder.
[0055] As will be apparent to those skilled in the art, the input
signal must be shifted 90 degrees to develop cos(X) to complete the
transform. The Hilbert filter is a well known method for achieving
a constant 90 degree phase shift over a wide range of frequencies.
The Hilbert filter is particularly well suited for implementation
in an FIR filter which is, in turn, well suited for DSP
applications. In consideration of the fact that Hilbert filters
require an odd number of filter coefficients, preferably a Hilbert
filter for producing the quadrature of the compressed audio signal
will employ at least 17 coefficients. As will also be apparent to
those skilled in the art, the incoming signal is shifted up one
octave by the above transform, precisely restoring the input signal
to encoder 300.
[0056] As with prior art frequency extenders, to make best use of
the bandwidth of a telephone line, it may also be desirable to
shift the frequency of the compressed signal up 250 Hz to achieve
good low frequency response across the phone line. If so desired,
this may be easily accomplished within the computer program for DSP
706 by processing the output of the transformation of either
encoder 100 or 300 according to the formula:
sin(X-250)=sin(X)*cos(250)+cos(X)*sin(250)
[0057] where:
[0058] sin(X) is the compressed audio; and
[0059] sin(X+250) is the signal delivered to the public
network.
[0060] At the receiving end, after digitization 206 or 406, but
prior to expansion 208 or 408, the 250 Hz offset may be removed
from the compressed audio according to:
sin(X)=sin(X+250)*cos(250)-cos(X+250)*sin(250)
[0061] As will be apparent to those skilled in the art, when
performed within the digital signal processor 706 (FIG. 7), the
shifting process described above is identical to that of prior art
frequency extenders. Preferably, the 250 Hz signal will be drawn
from a lookup table. Simultaneous generation of both sine and
cosine waves is then simply a matter of pulling two values, one for
sine, and the other for cosine, from the table with a fixed offset
between the pointers for each wave. It should be noted too that the
quadrature signal may be developed for the incoming audio signal
through a Hilbert filter as discussed hereinabove.
[0062] As will be apparent to those skilled in the art, compander
500 could include computer software to communicate with
conventional frequency extenders, as well as a mating compander
500. Acting as a frequency extender, compander 500 would simply
frequency shift uncompressed audio, as detailed above, up 250 Hz in
the encoding process, and down 250 Hz in the decoding process. Such
a device would be universal in the sense that, talent working for
multiple stations could use the device to send remote programming
to a station regardless of the local receiving equipment at the
station. Unprocessed audio could be sent to a station having no
special equipment. Frequency extended audio could be sent to a
station having only a prior art frequency extender. And frequency
companded audio could be sent to a station having a frequency
compander. As will also be apparent to those skilled in the art, it
would be possible, through spectral analysis of a test signal, such
as a 1 KHz sine wave, to distinguish the encoding scheme from among
the possible schemes. Upon determining the encoding scheme,
compander 500 could then automatically configure itself to operate
according to the compression or shifting scheme of the transmitting
device.
[0063] It is well known that various models and brands of older
frequency extenders were of questionable compatibility with each
other. The DSP of the inventive device may be programmed to
precisely tailor itself to any encoder or decoder at the other end
of the connection by analysis of a test signal, such as a 1 KHz
sine wave. As will be apparent to those skilled in the art, the
inventive system could thus be used to also implement a precision
frequency extender which avoids the problems associated with the
large number of passive components, the tolerances of such
components, and the costs and inaccuracies associated with analog
multipliers used in prior art frequency extenders.
[0064] As will also be apparent to those skilled in the art, the
companding process described herein could be repeated to achieve
any desired bandwidth, at least up to the point where the signal to
noise ratio becomes objectionable. In addition, in the FFT approach
described above, while the process was described with regard to
doubling the bandwidth, by a judicious selection of the sizes of
the frequency domain tables, it is possible to obtain virtually any
reasonable level of improvement in a single pass of the encoder and
decoder. Since the tables can be increased or decreased in size by
even a single location, fractional improvements in bandwidth are
even possible.
[0065] Yet another possibility of the present invention is that
both shifting and compression of the signal may be obtained by
manipulation of the frequency domain table. For example, the data
could be shifted up 250 Hz, as discussed above, simply by moving
the data in the frequency domain table up the appropriate number of
locations in the table. The 250 Hz shift of the compressed data
would occur automatically in the inverse FFT. Similarly, in the
expansion process, the data in the table would simply be shifted
down in the table by 250 Hz to remove the offset.
[0066] Thus, the present invention is well adapted to carry out the
objects and attain the ends and advantages mentioned above as well
as those inherent therein. While presently preferred embodiments
have been described for purposes of this disclosure, numerous
changes and modifications will be apparent to those skilled in the
art. Such changes and modifications are encompassed within the
spirit of this invention.
* * * * *